Showing papers on "Digital signal processing published in 1991"

Electronic still camera providing multi-format storage of full and reduced resolution images

Abstract: Electronic still imaging apparatus includes a digital signal processor (22) that transforms blocks of digital image signals derived from an image sensor (12) into sets of coefficient signals and encodes the coefficient signals into a stream of compressed signals. In addition, the digital processor generates reduced resolution image signals from the digital image signals and downloads both the compressed (high resolution) image signals and the reduced resolution image signals to a removable digital memory (24). By associating each high resolution image with its low resolution counterpart in a common multi-format image file, the image can be quickly accessed and a low resolution review image can be put up on a display device (116) without waiting for expansion and processing of the larger full resolution image.

The complex backpropagation algorithm

Abstract: The backpropagation (BP) algorithm that provides a popular method for the design of a multilayer neural network to include complex coefficients and complex signals so that it can be applied to general radar signal processing and communications problems. It is shown that the network can classify complex signals. The generalization of the BP to deal with complex signals should make it possible to expand the line of applications of this powerful nonlinear signal processing algorithm. >

Sonar Signal Processing

Abstract: Overview of digital signal processing conventional beamforming optimum detection and estimation in passive order practical implementations of optimum sonar detectors and estimators active sonar conventional passive sonar system matrix processing for sonar mapping matrix algorithms on to processor arrays recent developments glossary.

Digital signal processing system employing icon displays

Abstract: A method and system for generating icon displays representing digitized film scenes and scene processing operations, and manipulating the icon displays to initiate selected processing operations (such as transfers from one storage location to another) on the scenes. The invention can be embodied in a film scanning system having circuitry for digitizing scanned film images, for digitally correcting the digitized scanned images, and for inserting video sync information into the corrected images to convert the corrected images into a video signal. The system is also capable of inserting digital data in such a video signal, such as film in and out points, and data identifying a video storage location, and then storing the resulting video signal as a video format scene. In addition to generating a scene from a sequence of digitized film frames and related digital data, the system of the invention generates an icon representing the scene by compressing a selected one of the frames in the sequence. In response to a user command for information regarding a scene, the system displays the compressed scene image and other relevant scene data. The system also displays scene processing icons. In response to user selection of a scene icon and a scene processing icon, the system executes the corresponding processing operation on the corresponding scene.

Super resolution pitch determination of speech signals

Abstract: Based on a new similarity model for the voice excitation process, a novel pitch determination procedure is derived. The unique features of the proposed algorithm are infinite (super) resolution, better accuracy than the difference limen for F/sub 0/, robustness to noise, reliability, and modest computational complexity. The algorithm is instrumental to speech processing applications which require pitch synchronous spectral analysis. The computational complexity of the proposed algorithm is well within the capacity of modern digital signal processing (DSP) technology and therefore can be implemented in real time. >

Digital signal processing techniques for high accuracy ultrasonic range measurements

Abstract: Several digital signal processing (DSP) methods are analyzed and compared with respect to the expected errors for an ultrasonic range measurement arrangement. These include L1, L2 norms and correlation with different approaches for envelope extraction. The influence of different factors such as signal-to-noise ratio (SNR), sampling frequency, and digitizing resolution on measurement errors is analyzed using a synthetic approach through nearly 40000 simulations. Results show different performance levels involving accuracy, computing time, and cost for the studied methods, although all of them allow reduction of errors by several orders of magnitude. >

Modulation and coding for information storage

Abstract: Many of the types of modulation codes designed for use in storage devices using magnetic recording are discussed. The codes are intended to minimize the negative effects of intersymbol interference. The channel model is first presented. The peak detection systems used in most commercial disk drives are described, as are the run length-limited (d,k) codes they use. Recently introduced recording channel technology based on sampling detection-partial-response (or PRML) is then considered. Several examples are given to illustrate that the introduction of partial response equalization, sampling detection, and digital signal processing has set the stage for the invention and application of advanced modulation and coding techniques in future storage products. >

A radio system proposal for widespread low-power tetherless communications

Abstract: It is suggested that low-power digital radio as an access technology could be integrated into a local exchange network to provide a ubiquitous personal communications network (PCN). High-quality tetherless communications services that could be provided by such an exchange-network-based PCN are described. A possible low-power exchange access digital radio system for providing these exchange-network based PCS services is discussed. The radio system uses a spectrum-efficient time-division multiple-access (TDMA) architecture made possible by advanced digital signal processing techniques. Control of the frequency reuse system is described, and frequency spectrum needs are indicated. >

Signal Processing Algorithms

Abstract: Signal processing algorithms , Signal processing algorithms , مرکز فناوری اطلاعات و اطلاع رسانی کشاورزی

Multidimensional audio window management

Abstract: This paper proposes and organization of presentation and control that implements a flexible audio management system we call “audio windows”. The result is a new user interface integrating and enhanced spatial sound presentation system, an audio emphasis system, and a gestural input recognition system. We have implemented these ideas in a modest prototype, also described, designed as an audio server appropriate for a teleconferencing system. Our system combines a gestural front end (currently based on a DataGlove, but whose concepts are appropriate for other devices as well) with an enhanced spatial sound system, a digital signal processing separation of multiple sound sources, augmented with “filtears”, audio feedback cues that convey added information without distraction or loss of intelligibility. Our prototype employs a manual front end (requiring no keyboard or mouse) driving an auditory back end (requiring no CRT or visual display).

Carry-save architectures for high-speed digital signal processing

Abstract: Carry-save arithmetic, well known from multiplier architectures, can be used for the efficient CMOS implementation of a much wider variety of algorithms for high-speed digital signal processing than, only multiplication. Existing architectural strategies and circuit concepts for the realization of inner-product based and recursive algorithms are recalled. The two's complement overflow behavior of carry-save arithmetic is analyzed and efficient overflow correction schemes are given. Efficient approaches are presented for the carry-save, implementation of a saturation control. The concepts are extended and refined for the high-throughput implementation of decisiondirected algorithms such as division, modulo multiplication and CORDIC which have yet been avoided because of a lack of efficient concepts for implementation.

Signal Processing, Image Processing and Pattern Recognition

Abstract: A detailed exposition of the main areas of signal processing, this book is divided into three sections: one-dimensional signal processing and digital filters; two-dimensional signal processing and image processing; and pattern recognition. Among the more specific topics covered are: analog filters; discrete systems and signals; non-recursive filters; FFT; IIR design; quantization effects; and hardware and software design. There is also material on system stability, picture enhancement and restoration, and parallel processing methods, as well as a comprehensive treatment of syntactic methods, parsing and neural networks.

A self-steering digital microphone array

Abstract: A self-steering microphone array for teleconferencing in which the digitally implemented steering algorithm consists of two parts is presented. The first part, the beamforming, is based on known concepts. The second part, a novel voting algorithm, integrates elements of pattern classification and exploits temporal characteristics of speech signals. It accounts for perceptual criteria and the acoustic environment. A real-time implementation is outlined, and results are discussed. The results confirm that the proposed concept deals successfully with teleconferencing environments and that it yields substantially better performance than earlier concepts based on analog hardware. >

Two stage interpolation system

Abstract: A two stage interpolation system provides greater bandwidth for signals compressed and expanded by interpolation, such as video signals displayed in a zoom or enlarged mode. A finite impulse response filter (391) generates from a first signal of digital samples a second signal of digital samples representing signal points between the samples of the first signal. The first signal is delayed. The second signal and the delayed first signal are interleaved by a multiplexer (393) to produce a third signal of digital values having a sample density twice that of the first signal. A compensated variable interpolator (394) derives from the third signal a fourth signal of digital samples in which the frequency content of information represented by the first signal has been changed. In the case of enlarging the picture represented by a video signal, the video signal can be truncated in a FIFO line memory (395), prior to being delayed and processed by the finite impulse response filter (391). The compensated variable interpolator (394) can be controlled to provide selected ratios of compression or expansion.

Multi-frequency differentially encoded digital communication for high data rate transmission through unequalized channels

Abstract: A differentially encoded digital signal waveform is generated as a discrete time representation of a desired analog signal utilizing multi-frequency modulation techniques. The computational capability of present day, industry-standard microcomputers (57) equipped with a floating point array processor or digital signal processor chip is utilized to perform digital frequency encoding and compute both discrete Fourier transforms (31) and inverse discrete Fourier transforms (25) to provide a transmitter (51) and receiver (53) system utilizing suitably programmed microcomputers coupled by a communications channel.

The discrete Zak transform application to time-frequency analysis and synthesis of nonstationary signals

Abstract: Algorithms to compute coefficients of the finite double sum expansion of time-varying nonstationary signals and to synthesize them from a finite number of expansion coefficients are presented. The algorithms are based on the computation of the discrete Zak transform (DZT). Fast algorithms to compute DZT are presented. Modifications to the algorithms which increase robustness are given. >

Cathedral-III: Architecture-driven high-level synthesis for high throughput DSP applications

Abstract: The goal of this paper is to extend the synthesis of real time digital signal processing (DSP) algorithms towards the domain of high throughput applications. A novel architectural style specifically suited for this application domain is presented. Furthermore, a synopsis of a novel synthesis script typically oriented towards this architecture is described (architecture-driven synthesis). The emphasis in the script is on the design of the data-paths which are dedicated to the application, and special attention is paid to the memory synthesis problem. In this paper only the data-path related tasks, namely data-path partitioning and data-path definition, are discussed. The new methodology is demonstrated using an image processing application.

Subband coding of stereophonic digital audio signals

Abstract: The exploitation of left-right correlation in a subband code for stereophonic audio signals is investigated. A transform of left and right signals into decorrelated intensity and error signals is presented. Although this can be seen as the optimal exploitation of redundancy, it yields only marginal gain in bit rate. If the reduced phase-sensitivity of the human observer can be exploited by encoding only the intensity signal, a substantial gain can be obtained. Preliminary results of a stereo codec are promising: at 192 kb/s good coding results have been obtained. >

System and method for receiving and decoding global positioning satellite signals

Abstract: A system receives and demodulates spread spectrum positional signals such as those generated by a GPS satellite by frequency shifting the signals substantially to baseband and utilizing digital signal processing techniques. The digital signal processing techniques utilized can be implemented by a standard audio digital signal processor due to the circuit design. The spread spectrum signal is frequency shifted substantially to baseband, forming in-phase and quadrature components which are processed substantially in parallel. Pseudo-range, carrier phase and doppler frequency, and the underlying data are thereby derived.

C Language Algorithms for Digital Signal Processing

Abstract: The use of the C programming language to construct digital signal-processing (DSP) algorithms for operation on high-performance personal computers is described in a textbook for engineering students. Chapters are devoted to the fundamental principles of DSP, basic C programming techniques, user-interface and disk-storage routines, filtering routines, discrete Fourier transforms, matrix and vector routines, and image-processing routines. Also included is a floppy disk containing a library of standard C mathematics, character-string, memory-allocation, and I/O functions; a library of DSP functions; and several sample DSP programs. 83 refs.

30 Years Space Vectors, 20 Years Field Orientation, 10 Years Digital Signal Processing with Controlled AC-Drives, a Review (Part 1)

Abstract: SummaryThe dynamic interactions of AC-machines are far more complex than those of DC-machines and important quantities are not directly measurable. This has given rise to considerable difficulties in designing high performance AC-drive controls. About 20 years ago, they were overcome by the emergence or new methods of control, using moving frames of reference determined by the angular position or flux waves, hence called field orientation. Their application requires extensive on-line signal processing that can only be realised economically with microprocessors or special digital hardware.The paper presents a review of this evolution, which has matured in less than 2 decades from research studies to a universally accepted method for controlling AC-machines. The method of field orientation has proved to be well adaptable to all types of power converters and AC-machines; modifications and extensions to include self-tuning and adaptive features are possible. It now seems to be generally accepted that field or...

Representations of musical signals

Abstract: Fourteen articles by different authors describe digital audio and computer music systems made possible by advances in digital signal processing theory, hardware design, and programming techniques. They focus on models that combine time-domain and frequency-domain representations (grains, wavelets,

Digital signal processing: theory, applications, and hardware

Abstract: Designed for graduate students and signal processing practitioners with an introductory background in DSP, this new text gives representative coverage of advanced topics (orthogonal expansions, optimal filters, and two-dimensional DSP), and advanced aspects of familiar topics (fast transforms beyond the FFT, non-uniform sampling and quantization). Providing a self-contained blending of DSP theory, applications to speech and image processing, and state-of-the-art DSP hardware, "Digital Signal Processing" includes: introductory DSP concepts summarized in five appendixes; DSP filter algorithms - e.g.subband and median filters; least squares, optimal, and adaptive filters spectral estimation and deconvolution; speech and image processing applications; and DSP hardware realizations.

Image signal processing with digital filtering to minimize aliasing caused by image manipulation

Abstract: An image signal processing apparatus has an address generator that generates addresses for successive words of a digital input signal representing an input image to be manipulated. A computation circuit monitors the addresses for successive word sets which each comprise first and second words relatively horizontally spaced in the input image and a third word aligned vertically with the second word, and computes from each set both horizontal and vertical local scaling factors representing the extent of compression in the horizontal and vertical directions of a corresponding local area of the manipulated image. A bandwidth controller is responsive to the successive scaling factors to vary the horizontal and vertical bandwidths of a digital filter which two-dimensionally filters the input image, thereby to minimize aliasing that would be caused by the manipulation of the local areas. The computation circuit computes the scaling factors for each local area such that the horizontal factor corresponds to the length of a line that extends perpendicularly from a line passing through the positions of the second and third words in the manipulated image to the position of the first word and the vertical factor corresponds to the length of a line that extends perpendicularly from a line passing through the positions of the first and second words to the position of the third word.

Format conversion of digital video signals, integration of digital video signals into photographic film material and the like, associated signal processing, and motion compensated interpolation of images

Abstract: A system for integrating film material with a digital video signal employs a film scanner to produce a digital video signal from the source film, and a post production system for combining with that signal the input digital video signal Motion interpolated temporal compensation is employed at stages of frame rate conversion

Harmonic power flow determination using the fast Fourier transform

Abstract: Multichannel data acquisition systems and commercially available digital signal processing software packages make the determination of harmonic power flow possible provided the limitations of the analysis techniques are understood. The authors describe how data from a typical acquisition system can be processed using the fast Fourier transform and discuss possible errors (e.g. those due to leakage) and how to avoid them using techniques such as skewing correction and windowing. Guidelines on the practical application of the transform in analyzing measured data are presented. The analysis method has been used successfully to analyze data obtained at a traction (railway supply) substation which had a sixth-harmonic resonance caused by the interaction of a harmonic filter and the only supply system. >

System for separating speech from background noise

Abstract: A digital signal processing system applies an adaptive filtering technique to sequences of energy estimates in each of two signal channels, one channel containing speech and environmental noise and the other channel containing primarily the same environmental noise. From the channel containing primarily environmental noise, a prediction is made of the energy of that noise in the channel containing both the speech and that noise, so that the noise can be extracted from the mixture of speech and noise. The result is that the speech will be more easily recognizable by either human listeners or speech recognition systems.

Apparatus and method for reducing distortion in amplification

Abstract: Broadband amplifiers which are intended to be linear suffer from distortion and although many techniques have been devised to overcome this problem significant improvements are required especially at frequencies above 100 MHz. An error signal is formed from the output of the amplifier (4) by comparison in a subtractor (11) with its input and the error signal is combined in a coupler (17) with the amplifier output to reduce distortion. The present invention provides automatic amplitude and phase control of the signals forming the error signal by means of components (24 and 25) ina multi-channel input and in the error signal by means of components (13 and 14). These components receive control signals from feedback networks (18' and 20') which each process two of three input signals from: couplers (34, 36) before the subtractor (11), a coupler (22) which passes the error signal and a coupler (21) after the coupler (17). Preferably the two input signals used by a feedback network are used to form in-phase and quadrature signals by a 90° phase shifter (41) and mixers (42 and 43). These signals are each amplified (46, 47) and integrated (48, 49) to provide the control signals. The input signals from the feedback networks may be reduced in frequency and applied to a DSP which then provides the control signals.

Algebraic Methods for Signal Processing and Communications Coding

Abstract: This work aims to explore the links between digital signal processing and error-control codes, with the ultimate intention of making them two components of a unified theory or of making a large part of the theory of error-control codes a subset of digital signal processing. By studying the properties of the Fourier transform in an arbitrary field, a perspective emerges in which the two subjects can be unified. Because there are many fields and many Fourier transforms in most of these fields, the unified view also reveals a diverse set of mathematical tools, many of which yet to find an appropriate engineering application.